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versatica/mediasoup

7,247C++Audience · developerComplexity · 5/5Setup · hard

TLDR

A server-side media routing library (SFU) that handles forwarding audio and video streams in real-time group calls, available as a Node.js package and Rust crate, with WebRTC support but no built-in signaling or UI.

Mindmap

mindmap
  root((mediasoup))
    What it does
      Routes audio/video
      SFU media relay
      WebRTC support
    Tech Stack
      C++ worker
      Node.js
      Rust
    Use Cases
      Group video calls
      Live broadcasting
      RTP streaming
    Features
      Simulcast
      Congestion control
      Data channels
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Things people build with this

USE CASE 1

Build a group video calling app where audio and video are routed through your own server using WebRTC.

USE CASE 2

Create a one-to-many live streaming platform with adaptive quality so viewers on slow connections get a lower-resolution stream.

USE CASE 3

Stream audio and video via plain RTP from a device into a custom media pipeline without using a browser.

USE CASE 4

Handle large group calls efficiently by forwarding each participant's stream through a central SFU instead of peer-to-peer mesh.

Tech stack

C++Node.jsTypeScriptRustWebRTCRTP

Getting it running

Difficulty · hard Time to first run · 1h+

You must build your own signaling and room management logic, mediasoup only handles media routing, not participant coordination.

No license information was mentioned in the explanation.

In plain English

mediasoup is a server-side library for building real-time video and audio applications on the web. It works as what is called an SFU, or Selective Forwarding Unit. In a group video call, rather than having each participant send their video to every other participant directly (which becomes expensive as the group grows), an SFU sits in the middle, receives streams from each person, and forwards the right streams to the right places. mediasoup provides that routing layer. The library is available as a Node.js package and as a Rust crate, so you can use it in server-side code written in either language. The heavy media processing is handled by a C++ worker process running underneath, using highly efficient network I/O. On the client side there are thin TypeScript and C++ libraries that communicate with the server. By design, mediasoup handles only the media layer: getting audio and video from one point to another. It does not prescribe how participants should join a room, how they should signal to each other, or what a user interface should look like. You supply all of that yourself. This intentional minimalism means you can integrate mediasoup into almost any architecture without being forced into a particular pattern. Supported features include handling multiple audio and video streams over a single connection, simulcast (sending multiple quality levels of the same video so viewers on slower connections receive a reduced version), congestion control to adapt to changing network conditions, and data channel messaging. It works with both WebRTC and plain RTP as input and output, over UDP or TCP. Listed use cases include group video chat applications, one-to-many or few-to-many live broadcasting, and general RTP streaming scenarios.

Copy-paste prompts

Prompt 1
Using mediasoup in Node.js, show me how to create a room where two participants can send and receive video over WebRTC.
Prompt 2
Explain how to set up simulcast in mediasoup so viewers on slow connections automatically receive lower quality video.
Prompt 3
How do I configure mediasoup to accept plain RTP input instead of WebRTC, for example from an IoT camera?
Prompt 4
What signaling layer do I need to build on top of mediasoup to let users join and leave a video call, give me a minimal example?
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